VoIP in NREN Network : Research Education Network for

Uganda.
Masiga Brian Kevin Mugaya
2015/HD05/697U 2015/HD05/700U

ABSTRACT
Voice over IP has changed the face of collaboration and real time communication around the world. It
has acted as a catalyst for research collaborations and a platform for new teaching methods. VoIP is a
long term cost effective solution and it provides a reliable and simple platform for conducting meetings,
teaching and collaboration. In this paper, we are fronting a inter-institutional VoIP solution to the
Research and Education Network for Uganda (RENU) as a tool to foster collaboration among Institutions
of Higher Education in Uganda and Research Organisations. We illustrated the VoIP network architecture
for the RENU network using the Asterisk open source solution and described the server configurations of
the Asterisk machine. We further discussed the cost implications of the proposed solution in context of
Outbound VoIP calls to a PSTN network.

Keywords : Voice over IP (VoIP)

1.1 INTRODUCTION
Research and Education Network for Uganda (RENU) was established in 2006 by committee of
Institutional Vice Chancellors and Directors. Its main objective was to become a vehicle for facilitating
the human networking needed to boost intellectual output and trigger research led transformation
among institutions of higher education in Uganda. As part of the UbuntuNet Community funded by
Africa Connect Project, RENU has laid optic fiber cable across the entire country and through this fiber
backbone network, a number of institutions have been interconnected to foster the research growth in
the country. According to the RENU newsletter 2016 [1] , this has allowed a tariff free exchange of traffic
among the connected institutions at speeds of up to 1Gbps.
Well as the network backbone capacity is at 1Gbps for inter institution capacity, analysis of traffic usage
graphs suggests that none of the connected institutions have been able to fully utilize the available
bandwidth as shown in the figure 1. Figure 1 shows the inter-institutional traffic in RENU; a sample taken
from 17th to 18th January 2017. The maximum traffic hit was 100 Mbps that lasted a period of 30
minutes. Further analysis shows that most of the traffic exchanged on the network is traffic destined out
of the RENU network. The Figure 2 shows traffic sourced from Makerere University to a destination
within the RENU network. The maximum traffic hit was 300 kbps. The results shown were got from the
netflow monitoring tool within the RENU-NOC.

Figure 1: Inter-Institutional Bandwidth utilization graph

Figure 2: MUK Traffic to RENU Destinations

In this paper we suggest leveraging this available bandwidth to carry voice and video traffic across the
RENU Infrastructure. This would not only guarantee improved bandwidth utilization but also offers a
significant reduction in cost for voice and video communication across the member institutions.
Over the years, social software such as Voice over IP (VoIP) [2] has proven to be a catalyst for
collaboration amongst researchers. According to Humphrey et al [3], VoIP is a set of technologies that
allow voice and multimedia calls to be set up across an IP network. Meredith [2] explains further that,
using VoIP is much the same as talking on the phone only the signal is transmitted over the data network
rather than a phone line. Most NRENs in Europe have fully deployed VoIP in their networks and extended
it across NRENs in other countries across Europe in the GEANT network and this has shown tremendous
impact on communication amongst researchers. Miroslav et al states that 7 -10 % of voice calls become
free of charge and total telecommunications costs decreased by 48% per participant on average thanks
to the new voice services acquired. [4]

Institutions on the RENU network currently do not have access to an inter-institutional Telephony/VoIP
network, though some have on site VoIP services. Institutions such as Makerere University; Uganda
Christian University and Infectious Disease Institute have deployed VoIP services in their networks but
are locked up in communication within the bounds of their institutional network. This setup impedes on
the possible growth of collaboration among Ugandan researchers on the RENU network. The starting
idea for this project is to interconnect existing telephony system exchanges on institution premises
where available by using VoIP gateways to transform traditional telephony standards (for example ISDN
PRI connections) to VoIP, and then transfer data across the RENU IP network Infrastructure. For
institutions where traditional and VoIP telephony services are not available, we intend to provide servers
running asterisk SoftPBX software.

The rest of the paper is organised as follows:- the proposed network architecture is presented in section
1.2 ; section 1.3 presents the server configuration ; section 1.4 presents the cost implications; section 1.5
discusses the possible VoIP system challenges and section 1.6 presents the conclusion.
1.2 NETWORK ARCHITECTURE
In this section, we are proposing a network structure for the Inter-institutional VoIP system. The
proposed network structure extends to the Regional REN UbuntuNet Alliance that connects other NRENs
in Southern and Eastern Africa and a local Mobile Telecommunications Company (PSTN-Network).

a) PROPOSED VOIP NETWORK STRUCTURE

Figure 3: The Proposed VoIP Network Prototype

The proposed design considers 4 out of the over 50 institutions currently connected to the RENU
backbone network. Each of these institutions is guaranteed up to 1Gbps speeds across the RENU
infrastructure. It’s part of this bandwidth that we hope to leverage for inter-institutional VoIP and Video
Traffic.
From the proposed structure as shown in figure 3, VoIP calls from individual institutions arrives first at
the RENU_DATA_CENTER before they can be redirected to another institution via the RENU backbone
network or to a public switched telephone network service provider. These calls may also be redirected
to other NRENs that have functional VoIP networks and are part of the greater UbuntuNet Alliance
community.

b) CONNECTION OF AN INSTITUTION

Figure 4: Connecting Institutions

The proposed physical connection of an institution is as shown in the figure 4. Considering a scenario like
Kyambogo University which still has predominantly legacy telephone systems within its network, the
calls will have to be transferred from a legacy PBX to a VoIP gateway using ISDN Primary rate interfaces.
The gateway will convert the digital voice traffic into IP traffic and will transport it across the RENU
backbone using SIP signaling. Digium the company responsible for asterisk software has a number of
these gateways which we intend to use in our network.
Institutions like Uganda Christian University and Makerere University which have an existing VoIP
network need not purchase gateways. Rather can simply be connected to the RENU Asterisk server via
their existing VoIP IP PBX using SIP or H.323 signaling Protocols.

c) PEERING CONNECTIONS
The Asterisk server used at the RENU data centre can be used to accept peering connections with the
connected university telephony systems. This can be achieved using SIP signaling protocol or the PjSIP
protocol which has been used since the release of asterisk 13 to allow for communication between
networks. For our proposed design SIP peering will be used between the RENU data centre and
connected institutions and will also be used to connect to outside NRENs through a peering session on
the UbuntuNet server.

d) PSTN CONNECTIONS
In order to improve user acceptability of the proposed network, it’s important to offer the users a
possibility of placing calls across traditional telecommunication networks. This can only be achieved by
setting-up connections with any of the telecommunication companies. Connections to the service
provider network will be done using ISDN/PRI. The number of simultaneous calls that can be placed
across these connections will depend on what is deemed acceptable by RENU.

1.3 COST IMPLICATION

The proposed network will offer free Intra-RENU and Outbound (On-UbuntuNet) voice and video calls
allowing for improved research collaboration by taking advantage of underutilized bandwidth on the
RENU Network.
Calls outbound RENU and routed through mobile telecommunication networks will be charged as per
agreement between RENU, the Member Institutions and the selected mobile service provider.
In the figures below we look at current prices as per PSTN connection across select mobile service
providers
Figure 4: PSTN connection pricing
Based on the figures above, we can see that the price for a connection to a PSTN service provider
depends on;- a) The number of ISDN-PRI/BRI channels; b) The transmission media; c) The destination of
the call. Africell offers the E1 connection service over microwave for a one off installation fee of
1,668,381 Ugandan Shillings with an equipment fee of 3,370,811 Ugandan Shillings. You are then
required to pay a monthly E1 rental fee of 300,000 Ugandan Shillings.
One E1 equates to 32 Channels (E1-PRI) with 30 dedicated channels for voice and data transmission and
2 channels for carrying signaling information. What this means, is that over a single E1 connection you
are only allowed 30 simultaneous voice connections for a fee of 300,000 Ugandan Shillings across the
Africell Network. Airtel has a provision of offering the E1 service over fibre. The installation fee is 100
USD tax exclusive and a monthly E1 rental fee of 300,000 Ugandan Shillings. It’s important to note that
the stated prices do not include the different prices charged on per call basis with international calls
commanding a higher charge.
Calls outbound RENU and destined to other NRENs will be taken up in the institutional monthly
international capacity cost. Table 1 below shows the breakdown

ITEM COST (US Dollars)

Dell Server - 150 GB RAM, 5 TB disk space 7000

Desk VoIP Phones 100 per handset

International Bandwidth Cost 95 per Mbps

PSTN - Africell 82 per month for 30 channels

PSTN - UTL 62 per month for 15 channels

PSTN -AIRTEL 82 per month for 30 channels

Table 1: Breakdown of the VoIP Installation Cost
1.4 SERVER CONFIGURATIONS
In this section, we present the basic configurations that are needed on a server in order to setup an
asterisk soft PBX.
For this proposal, we used linux Ubuntu-16.04.1-server operating system. The server OS was installed on
a virtual machine in Vmware Workstation 12.
Below, we list the steps followed in order to set-up the SoftPBX.
1) Update the Linux server OS to ensure any outdated dependencies and programs are upto date.
2) Download the latest asterisk version release. (in our case we downloaded asterisk 13.13.1, this is
a long term support version). The download will be in a tarball format.
3) Untar the downloaded asterisk application into a location of your choice.
4) Using apt-get package manager, download and install any dependencies that will be required for
asterisk to run without any errors. (a list of these can be forward on the asterisk.org website.)
5) Using the change directory command move into the asterisk directory.
6) Next step is to install asterisk. To do this, we had to first of all have to set-up all the directories
required to install asterisk. This we did by issuing the ./configure command. Once the command
was run successfully we then run the make menuselect command. This allowed us to choose the
applications that we needed in our PBX among other things. Once we were done with the
menuselect command, we then issued the make and make install command to compile the
asterisk code, if ok then proceed to install the asterisk application.
7) Finally we ran the make config command to build initialization scripts. These ensure that at all
times if the server ever reboots, it starts up with your asterisk pbx running.
So after the steps above, your asterisk PBX should be up and running. The next step is to set it up as per
requirement.
For our case the PBX is supposed to manage outbound calls and calls inbound within the RENU network.
As per our design, calls within RENU are managed using the SIP protocol.
Figure 5 below shows a basic SIP configuration example from which our proposed network can be built.
 Figure 5: Basic SIP_Configuration

The configuration is as per the SIP RFC 3261. This document clearly explains how this protocol can be
used to setup and manage VoIP calls. Within asterisk, SIP is configured in the SIP.CONF file. It’s important
to note that SIP will provide the platform and guidelines for our calls but it does not set up the dial plan
i.e the numbers, or the users allowed to make calls on our PBX system.
This is done in the extensions.conf file. Within this we are now able to setup the database of users
allowed to register/use the phone system.
Figure 6 below shows a basic dial plan for our RENU project.
 Figure 6: Basic DialPlan

Within asterisk, each dial plan is linked to the SIP.Conf file via the context. In this case the context is
Softphones. The dial plan indicates the extension number; the priority; application; protocol and the sip
registered username. The priority number indicates the order to be followed within a particular context
once a particular extension is dialed.

1.5 POSSIBLE SYSTEM CHALLENGES

In this section, we present the possible challenges in deployment and maintenance of the RENU VoIP
system. Some of the challenges presented are as a result of nature and state of the VoIP service.

Maintenance of the VoIP System

Most IT systems deployed in Institutions of Higher Education in Uganda face the maintenance problem
that ranges from the skill set needed by the IT personnels to run the system; to the cost of maintenance
in terms of System Upgrades of the existing software and purchase of the required system hardware
incase of breakdown or performance related issues associated with hardware.
The proposed RENU VoIP system is no exception to this problem as it requires that most of the IT
personnels in the institutions have a skill level to provide basic administration and troubleshooting of the
system. This skill set is vital for usability requirements such as system uptime and system continuity. In
the proposed VoIP system design, each Institution will have a VoIP server that the institution manages
and administers. The institution will also carry the burden of hardware replacement in case of a
breakdown and when new hardware parts are required to enhance service quality.
Billing of VoIP services
In our proposed VoIP system design, extended VoIP services to destinations outside the RENU network
will be billed by RENU on a per institutional basis. Basically, VoIP calls outbound the RENU network and
routed through the PSTN network will be charged as per agreement between RENU, the Member
Institution and the selected mobile service provider. RENU will have to set up an integrated billing system
for this purpose and differentiate VoIP calls traffic between institutions .

Quality of the Institutional Networks.

VoIP as a service is sensitivity to Jitter and Packet drops in a network. [5] Networks with high jitter and
packet drops affect the quality of VoIP calls and video services in any VoIP system. With this in mind, the
quality of network equipment deployed in the institutional Local Area Networks (LAN) becomes pivotal
to the success of the proposed VoIP system. RENU is currently playing a vital role in building resilient and
efficient networks in member institutions by conducting Direct Engineering Assistance (DEA) workshops
at individual Institutional sites coupled with Campus Network Design workshops.

Security of the VoIP System

VoIP as any other internet service is exposed to major security concerns over the internet. VoIP as a
service is exposed to Denial of Service attacks (DOS); eavesdropping on calls using SipTap [6]; telephony
spam [7] and patching problems. It therefore becomes critical that Network security audits are
performed on a per institutional basis coupled with deployment of Firewalls and Intrusion Detection
Systems. That said, the cost implications of the implementation increase.

1.6 CONCLUSION
In conclusion, we believe that Inter-Institutional VoIP solution provides a simple and cost effective way of
communication and collaboration. It also takes advantage of the available unused bandwidth capacity on
the RENU -backbone interconnecting institutions. The proposed VoIP network architecture and design
gives autonomy to the institutions to manage and administer their own VoIP deployment thus the
solution is easily scalable and efficient. The cost implications discussed in the paper show that in overall
a VoIP System is a much cheaper option that allows for; cheaper internal calls and reduces overall
maintenance cost all while offering much more services compared to traditional telephony services.
REFERENCES
[1] Beyond Connectivity , RENU Newsletter 2016. Published on the RENU website [www.renu.ac.ug].

[2] Social Software in Libraries, 2007. Meredith G Frakas.

[3] Collaborating through Real-Time VoIP and Visual Search, 2008. Cathal Hoare and Humphrey
Sorensen.

[4] Set of IP Telephony Best Practices in National Research Networks in EU, March 2011. Miroslav
Voznak; Ulf Tigerstedt; Branko Radojevic; Michal Halas; Miguel Duarte.

[5] Understanding Jitter in Packet Voice Networks. February 02, 2016. Cisco Systems, Document ID
18902.

[6] Eavesdropping and interception security hole and its solution over VoIP service, December 2014.
Aditya Dakur and Shruthi Dakur.

[7] On Spam over Internet Telephony (SPIT) Prevention, August 2008. Juergen Quittek; Saverio Niccolini;
Sandra Tartarelli and Roman Schlegel.